[Freeswitch-users] pocketsphinx and event socket
Hello, I have tried the pizza demo and did it get to work so far. However I would like to use pocketsphinx through event socket. I saw in the wiki that there is a chapter for Speech Synthesis Commands: http://wiki.freeswitch.org/wiki/Mod_commands#Speech_Synthesis_Commands However this is empty. Also http://wiki.freeswitch.org/wiki/ASR didn't give me a hint. As I am not a Java programmer, it's hard for me to determine how the pizza demo actually works. Anybody has a sample how he did it e.g. in Php/Perl or so? (I am working with Ruby) Or back to the basics: Is it possible to use pocketsphinx through event socket? Best regards Peter ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] SIP server? PBX vs. softswitch?
Hello Even though I successfully set up an Asterisk voice server, I'm no telecom expert, and would like some clarification about the following things: - What is an SIP server as opposed to a IP PBX? - What is the different between a PBX like Asterisk and a softswitch? Thank you. ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] freeswitch with celt on windows
Hi. I was wondering if it is possible to compile freeswitch with the celt codec on windows? I have been able to compile celt for windows but when i compiled freeswitch i did not see the mod_celt in the solution explorer. If this is possible could someone point me in the right direction. Thank you. ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] pocketsphinx and event socket
You can accomplish this here is an example using ESL in perl http://fisheye.freeswitch.org/browse/FreeSWITCH/libs/esl/perl/server3.pl?r=12344 /b On Feb 28, 2009, at 6:18 AM, Peter P GMX wrote: Or back to the basics: Is it possible to use pocketsphinx through event socket? ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] Using OpenZAP + FXO card just to get CID info?
Hello I'd like to write a single-host CRM application, so I need to get the CallerID information when a call comes in. I don't actually need a PBX/softswitch. The user will have the FXO cards and a phoneset connected on the same line, and will pick up the phone once the CRM application has picked up the CID info and popped up a dialog box, etc. Is it possible to just use OpenZAP to get this information, or must I install Freeswitch and provide an IP phone as well? Thank you. ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Using OpenZAP + FXO card just to get CID info?
OpenZAP is just a module accessing the card, you need to use it within freeswitch. Then, you can use event socket to get the callerid. On 28-Feb-09, at 11:30 AM, Fred wrote: Hello I'd like to write a single-host CRM application, so I need to get the CallerID information when a call comes in. I don't actually need a PBX/softswitch. The user will have the FXO cards and a phoneset connected on the same line, and will pick up the phone once the CRM application has picked up the CID info and popped up a dialog box, etc. Is it possible to just use OpenZAP to get this information, or must I install Freeswitch and provide an IP phone as well? Thank you. ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] freeswitch with celt on windows
Someone just needs to do the work of adding it to the build... I thought Carlos did this already... Are you on SVN Trunk? /b On Feb 28, 2009, at 10:01 AM, e schmidbauer wrote: Hi. I was wondering if it is possible to compile freeswitch with the celt codec on windows? I have been able to compile celt for windows but when i compiled freeswitch i did not see the mod_celt in the solution explorer. If this is possible could someone point me in the right direction. Thank you. ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] SIP server? PBX vs. softswitch?
It depends on how you look at it... most will say there is no difference... but last I checked you usually don't run heavy apps on a softswitch. FreeSWITCH can be everything from softphone to softswitch and everything in between including PBX. The default config comes configured as a PBX. /b On Feb 28, 2009, at 9:47 AM, Fred wrote: Hello Even though I successfully set up an Asterisk voice server, I'm no telecom expert, and would like some clarification about the following things: - What is an SIP server as opposed to a IP PBX? - What is the different between a PBX like Asterisk and a softswitch? Thank you. ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] freeswitch with celt on windows
i used the nightly snapshot... http://files.freeswitch.org/freeswitch-snapshot.tar.gz On Sat, Feb 28, 2009 at 11:34 AM, Brian West br...@freeswitch.org wrote: Someone just needs to do the work of adding it to the build... I thought Carlos did this already... Are you on SVN Trunk? /b On Feb 28, 2009, at 10:01 AM, e schmidbauer wrote: Hi. I was wondering if it is possible to compile freeswitch with the celt codec on windows? I have been able to compile celt for windows but when i compiled freeswitch i did not see the mod_celt in the solution explorer. If this is possible could someone point me in the right direction. Thank you. ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] SIP server? PBX vs. softswitch?
Where this is distinguished, it is not directly at the level that user's experience the end result. In the case of what is called a softswitch, one answer is found in organizations like the ISC (International Softswitch Consortium) and vendors who built products around their architecture recommendations. These systems tend to be very complex and componetized, where basic functionality operates in self-contained components that then interact with the whole through defined open standards and network protocols, such as SIP. The primary reason for ISC-style architectures is a result of proprietary development, where code and internal operations cannot be shared or modified. Hence, by breaking up functionality into subcomponents, it is possible to replace a component subsystem as a whole while retaining the interfaces. A perfect example is call forwarding. In a traditional proprietary (ISC-model) softswitch, call forwarding would be an entirely separate self-contained proprietary feature server interacting over SIP. If someone wants to create a different call forwarding behavior, one slips in an alternate server. By contrast, it is far easier in an open source/free software PBX to simply modify the feature code that implements call forwarding directly to create new and specialized versions of that feature. Hence, you do not find or have need for micro-services for tiny features in pbx software that originated as open source and free software or that did not follow the path of proprietary architectures, such as Bayonne, Asterisk, or FreeSwitch. A perfect example of a traditional softswitch architecture is SipX, which originated as a proprietary VoIP pbx codebase. However, even at this point, such distinctions I think are still somewhat artificial, as Brian suggests. What does distinguish architectures that may be relevant to end users is whether a IP-PBX solution operates as a B2BUA (back-to-back user agent) or not. A pure B2BUA solution is one where all media as well as signalling goes directly through the central PBX switch. A perfect example of this is how Asterisk traditionally works. This makes it very easy to adapt and connect multi-protocol endpoints, to convert media formats for endpoints who do not have common codecs, etc, since all media endpoints talk to the switch rather than each other. However, since all media goes through a central point, the scalability of such systems can often become compute-bound, and extra latency is induced. A pure network solution by contrast has all media connect directly peer to peer by the user agent endpoints, and the pbx really only handles and coordinate independently operating endpoints through signalling. This often requires separate servers for gateways to the PSTN or other protocols. But it does offer better latency and scalability, and the ability to provide end-to-end media security, such as when using ZRTP. This difference, between B2BUA and non-B2BUA, is I think far more relevant today than traditional classifications such as IP-PBX, softswitch, SIP Server, etc. Brian West wrote: It depends on how you look at it... most will say there is no difference... but last I checked you usually don't run heavy apps on a softswitch. FreeSWITCH can be everything from softphone to softswitch and everything in between including PBX. The default config comes configured as a PBX. /b On Feb 28, 2009, at 9:47 AM, Fred wrote: Hello Even though I successfully set up an Asterisk voice server, I'm no telecom expert, and would like some clarification about the following things: - What is an SIP server as opposed to a IP PBX? - What is the different between a PBX like Asterisk and a softswitch? Thank you. ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org begin:vcard fn:David Sugar n:Sugar;David org:GNU Telephony email;internet:dy...@gnutelephony.org tel;work:+1 609 465 5336 url:http://www.gnutelephony.org version:2.1 end:vcard ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] Orginate: getting status of call fail
Hi Guys,
I've been running a test script written in lua which now works very well
thanks to Anthony's fix to stream file.
Right now I'm using an event socket to initiate the call and passing the
name of the script along with originate thus:
$dialstring = originate
{ignore_early_media=true,origination,originate_timeout=25}sofia/gateway/
Mygw/phonenum 'lua(helloworld.lua )';
$result = $obj -bgapi_command($dialstring);
The script gets fired (it would appear) on answer. However, if the
number is invalid , timed out or was busy, I'm not sure the script gets
executed or am I wrong?
I want to be able to fire an event back on what happed to the call in
the event that it failed for whatever reason.
I know I can simply call the originate and pass the number as an
argument and execute the dial within the script but I'm led to believe
that's not very efficient, or am I completely wrong?
Looking for the most FS friendly way here
Regards,
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[Freeswitch-users] Problems loading mod_spidermonkey_curl
I'm getting the following error when I try to load the mod_spidermonkey_curl module. I didn't get any errors when I compiled it. I also tried --without-libcurl but I got the same result. Any help would be appreciated. freeswi...@internal load mod_spidermonkey_curl -ERR [module load file routine returned an error] 2009-02-28 14:17:54 [CRIT] switch_loadable_module.c:839 switch_loadable_module_load_file() Error Loading module /usr/local/freeswitch/mod/mod_spidermonkey_curl.so **/usr/local/freeswitch/mod/mod_spidermonkey_curl.so: undefined symbol: mod_spidermonkey_curl_module_interface** freeswi...@internal version FreeSWITCH Version 1.0.3 (exported) ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Problems loading mod_spidermonkey_curl
please open up spidermonkey.conf.xml and add it to the load there... its a sub module of mod_spidermonkey so you can't load it at the CLI /b On Feb 28, 2009, at 1:33 PM, David Dan wrote: I'm getting the following error when I try to load the mod_spidermonkey_curl module. I didn't get any errors when I compiled it. I also tried --without-libcurl but I got the same result. Any help would be appreciated. freeswi...@internal load mod_spidermonkey_curl -ERR [module load file routine returned an error] 2009-02-28 14:17:54 [CRIT] switch_loadable_module.c:839 switch_loadable_module_load_file() Error Loading module /usr/local/ freeswitch/mod/mod_spidermonkey_curl.so **/usr/local/freeswitch/mod/mod_spidermonkey_curl.so: undefined symbol: mod_spidermonkey_curl_module_interface** ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Problems loading mod_spidermonkey_curl
That did it. Thank you On 2/28/09, Brian West br...@freeswitch.org wrote: please open up spidermonkey.conf.xml and add it to the load there... its a sub module of mod_spidermonkey so you can't load it at the CLI /b On Feb 28, 2009, at 1:33 PM, David Dan wrote: I'm getting the following error when I try to load the mod_spidermonkey_curl module. I didn't get any errors when I compiled it. I also tried --without-libcurl but I got the same result. Any help would be appreciated. freeswi...@internal load mod_spidermonkey_curl -ERR [module load file routine returned an error] 2009-02-28 14:17:54 [CRIT] switch_loadable_module.c:839 switch_loadable_module_load_file() Error Loading module /usr/local/ freeswitch/mod/mod_spidermonkey_curl.so **/usr/local/freeswitch/mod/mod_spidermonkey_curl.so: undefined symbol: mod_spidermonkey_curl_module_interface** ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Sent from my mobile device ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
